Scalable transcoding for streaming audio

ABSTRACT

Systems and techniques for capturing audio and delivering the audio in digital streaming media formats are disclosed. Several aspects of the systems and techniques operate in a cloud computing environment where computational power is allocated, utilized, and paid for entirely on demand. The systems and techniques enable a call to be made directly from a virtual machine out to a Public Switch Telephone Network (PSTN) via a common Session Interface Protocol (SIP) to PSTN Breakout service, and the audio to be delivered onward to one or more Content Delivery Network (CDN). An audio call capture interface is also provided to initiate and manage the digital streaming media formats.

TECHNICAL FIELD

This disclosure relates generally to streaming data, and moreparticularly to scalable transcoding for streaming audio.

BACKGROUND

Generally, a communications platform that transforms audio into anintegrated streaming audio webcast is known in the art. End-users areable to participate in the audio webcasts from anywhere on the Internetusing just a standard web browser, with the audio being streamed to theuser. Streaming media is a type of Internet content that can be playedwhile still in the process of being downloaded. A user's computer canplay the first packet of an audio stream, decompress the second, whilereceiving the third. As such, an end-user can listen to a webcastwithout waiting until the end of content transmission. Streaming mediaquality can vary widely according to the type of media being delivered,the speed of the user's Internet connection, network conditions, contentencoding, and the format used.

Typically, the media's server, client, and production and encoding toolsdeveloped by a streaming software vendor are collectively referred to asa format, Streaming media encoded in a particular format is provided bythat format's media server and can be replayed using that format'sclient. Media clients are also often referred to as ‘players’, andtypically exist as plug-ins to Web browsers. Example players include,but are not limited to, Windows Media®, Real Player®, Apple QuickTime®and Adobe Flash®.

Audio webcasts have been used for several years by companies tocommunicate with investors and security analysts. For example, on Oct.23, 2000, the Securities and Exchange Commission (SEC) adoptedRegulation FD (Fair Disclosure) which provides that when an issuer, orperson acting on its behalf, discloses material nonpublic information tocertain enumerated persons (in general, securities market professionalsand holders of the issuer's securities who may well trade on the basisof the information), it must make public disclosure of that information.Various companies have conformed to Regulation FD by disclosing materialnonpublic information to the public using audio webcasts.

Turning now to FIG. 1, a typically audio webcast will be discussed. Tobegin, at step 10, a customer wishing to initiate a webcast contacts acall provider to schedule an event. At step 12, the call provider, whotypically operates an event registration system for scheduling theevent, enters the event information into the registration system. Then,at step 14, the call provider confirms the event information and sendsthe customer a provider Web site address, an event identifier, and auser name and password to use to initiate the conference call. At step15, either the vendor and/or call provider transmits invitation messagesto prospective end-users. The messages are typically included in ane-Mail and include the event identifier sent to the customer, as well asa link to a content distribution Web site. At step 16, the prospectiveend-users receive the notification. Lastly, at step 18, to access theevent, an end-user selects the link included in the e-Mail (or enters aURL manually) to launch his or her browser's media player to listen tothe event. To connect and listen to an event, the end-user typicallyrequires a computer with a hardware sound card and Internet connection,an Internet browser (Internet Explorer or Netscape Navigator, or thelike), streaming media player (e.g., Windows Media Player, RealPlayer orthe like) and the Web site address of the event. At the Web site addressof the event, the end-user may enter the event identifier, user name (ifrequired) and password (if required) to access the event. Of course, oneor more of the above-described steps can be carried out in a differentmanner. At any one point in time, several hundred individuals mayparticipate in an audio webcast and multiple audio webcasts, asdescribed in connection with FIG. 1, can occur simultaneously todisseminate material nonpublic information and other information.

Traditional webcast systems, however, have several deficiencies. Forexample, these systems typically operate on a single computer server,which represents a single point of failure and limits scalability, i.e.,the number of users that can listen to the audio of the event. Moreover,the prior art systems require advanced setup for the content streamsrequiring a significant investment in both computer and telephonyinfrastructure equipment. Such systems and resources required include,without limitation, racks of telephony equipment, media encoders,storage, network connectivity, and the like. Moreover, thisinfrastructure is required to be maintained twenty four (24) hours,seven (7) days a week for three hundred and sixty five (365) days inreadiness for service. Furthermore, the capacity of this infrastructureneeds to exceed the highest possible peak of demand, even if averagedemand only utilizes a fraction of the equipment. As a consequence,prior art webcast systems require physical production facilities thathave inherent cost and scaling issues.

These and other problems of prior art webcast systems are addressed bythe present invention.

SUMMARY

Systems and techniques for capturing audio and delivering the audio indigital streaming media formats are disclosed. Several aspects of thesystems and techniques operate in a cloud computing environment wherecomputational power is allocated, utilized, and paid for entirely ondemand. The systems and techniques enable a call to be made directlyfrom a virtual machine out to a Public Switch Telephone Network (PSTN)via a common Session Interface Protocol (SIP) to PSTN Breakout Service,and the audio to be delivered onward to one or more Content DeliveryNetwork (CDN). An audio call capture interface is also provided toinitiate and manage the digital streaming media formats.

Various aspects of the invention relate to streaming and encoding audiousing a cloud computing environment. For example, according to oneaspect, a method of streaming information includes establishing a set ofconnections between a first PSTN breakout service and a set of machines.A first machine in the set of machines is a first virtual machineinstantiated in a first cloud computing environment, the first PSTNbreakout service being connected to a conference call. The method alsoincludes receiving a digital data stream in at least one machine in theset of machines, the digital data stream having been encoded accordingto a first encoding protocol, encoding the digital data stream togenerate an encoded digital data stream, the encoding being doneaccording to a second encoding protocol, and transmitting the encodeddigital data stream to a content delivery network (CDN).

In one embodiment, the method includes receiving the digital data streamover one of the set of connections between the first PSTN service andthe first virtual machine, encoding the digital data stream on the firstvirtual machine, and transmitting the encoded digital data stream fromthe first virtual machine to the CDN. The method may also includereceiving the digital data stream at a second virtual machine, encodingthe digital data stream on the second virtual machine, and transmittingthe encoded data stream from the second virtual machine to the CDN.

The second virtual machine may be instantiated in the first cloudcomputing environment or a second cloud computing environment that isdifferent from the first cloud computing environment. In addition, themethod may include instantiating the first virtual machine in responseto a request from a call control interface module. In one embodiment,the method includes instantiating the second virtual machine from atleast one of the call control interface module and the first virtualmachine.

In one embodiment, the method includes receiving the digital data streamat the second virtual machine from the first virtual machine. In anotherembodiment, the method includes receiving the digital data stream at thesecond virtual machine from the first PSTN breakout service. In yetanother embodiment, the method includes receiving the digital datastream from a second PSTN breakout service, the second PSTN breakoutservice being connected to the conference call.

The method may include transmitting the digital data stream to thesecond virtual machine from the first PSTN breakout service upon atleast one of a failure and delay in one of the data connections betweenthe first data connection and the first virtual machine. The digitaldata stream may be received using a speech codec, such as G.722, G.719,G.711 or SPEEX.

In one embodiment, the method includes providing a session initiationprotocol (SIP) application in each of the first and second virtualmachines for receiving and managing the digital data stream in the firstencoding protocol, and providing at least one audio encoder for encodingthe digital data stream into the second encoding protocol.

The method may also include providing an operating system (OS) audio busfor transmitting the digital data stream in each of the first and secondvirtual machines, the OS audio bus being a software audio driver. In oneembodiment, the software audio driver is a Windows Driver Model (WDM)audio driver. The method may also include encoding the digital datastream into a Windows Media Audio (WMA) file.

The method may further include providing a plurality of audio encodersfor encoding the digital data stream into at least one third encodingprotocol different from the first encoding protocol and the secondencoding protocol, encoding the digital data stream using the thirdencoding protocol, and transmitting the encoded digital data streamaccording to the third protocol to the CDN. In one embodiment, a firstencoder of the plurality of audio encoders is a Windows Media encoderand a second encoder of the plurality of audio encoders is a Flash Mediaencoder. The method may also include encoding the digital data streaminto a Flash Media audio file.

The method may also include providing an audio server in each of thefirst and second virtual machines, the audio server transmitting theencoded digital data stream to the CDN. In one embodiment, the audioserver is a flash media server and the encoded digital data stream is aflash media file.

In yet another embodiment, the method further includes generating, fromthe SIP Application, an MP3 audio file from the received digital datastream in the first encoding format, and storing the MP3 audio file in adata store associated with each of the first and second virtualmachines. The method may further include audio scrubbing the MP3 audiofile.

Systems, methods, as well as articles that include a machine-readablemedium storing machine-readable instructions for implementing thevarious techniques, are disclosed. Details of various implementationsare discussed in greater detail below.

Additional features and advantages will be readily apparent from thefollowing detailed description, the accompanying drawings and theclaims.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is an example method of initiating and conducting an audioconference cast in accordance with the prior art.

FIG. 2 is a schematic of an example transcoding system according to oneembodiment of the present invention.

FIG. 3 is a schematic of virtual machines included in the exampletranscoding system of FIG. 2.

FIG. 4 illustrates an example method of capturing conference call audioand delivering the conference call audio in streaming media formatsaccording to one embodiment of the present invention.

Like reference symbols in the various drawings indicate like elements.

DETAILED DESCRIPTION

Turning now to FIG. 2, an example of a suitable computing system 20 forcapturing conference call audio via a Public Switch Telephone Network(PSTN) 52 and delivering the conference call audio in streaming mediaformats to one or more content distribution networks (CDN) 54 isdisclosed. The computing system 20 is only one example and is notintended to suggest any limitation as to the scope of use orfunctionality of the invention. The computing system 20 should not beinterpreted as having any dependency or requirement relating to any oneor combination of illustrated components.

For example, the present invention is operational with numerous othergeneral purpose or special purpose computing consumer electronics,network PCs, minicomputers, mainframe computers, laptop computers, aswell as distributed computing environments that include any of the abovesystems or devices, and the like, at least some of which may beconfigured in cloud computing environments.

The invention may be described in the general context ofcomputer-executable instructions, such as program modules, beingexecuted by a computer. Generally, program modules include routines,programs, objects, components, data structures, loop code segments andconstructs, etc. that perform particular tasks or implement particularabstract data types. The invention can be practiced in distributedcomputing environments where tasks are performed by remote processingdevices that are linked through a communications network. In adistributed computing environment, program modules are located in bothlocal and remote computer storage media including memory storagedevices. Tasks performed by the programs and modules are described belowand with the aid of figures. Those skilled in the art can implement thedescription and figures as processor executable instructions, which canbe written on any form of a computer readable media.

As shown in the FIG. 2 example, the system 20 includes a server device22 configured to include a processor 24, such as a central processingunit (‘CPU’), random access memory (‘RAM’) 26, one or more input-outputdevices 28, such as a display device (not shown) and keyboard (notshown), and non-volatile memory 30, all of which are interconnected viaa common bus 32 and controlled by the processor 24.

In one embodiment, the server device 22 is in operative communicationwith a plurality of cloud computing environments 40, 42 collectivelyconfigured with a plurality of virtual machines 44A-D. The server device22 provides a control plane (e.g., job control) via link 58 and includesa scheduling module 34 that is used to schedule events, such as audioconference calls, and an initiation module 36 for instantiating thevirtual machines 44A-D, network operation center (NOC) user interfaces46A, 46B, and data stores 56A, 56B in each cloud environment 40, 42.

The data stores 56A, 56B shown in FIG. 2 are a repository that storesstate information concerning each virtual machine 44A-D, respectively,operating in each respective cloud environment 40, 42. In oneembodiment, the data stores 56A, 56B are relational databases configuredin each cloud computing environment 40, 42. Database replication isimplemented across each of the databases 56A, 56B. In anotherembodiment, the data stores 56A, 56B are directory servers, such as aLightweight Directory Access Protocol (‘LDAP’) that are replicatedacross each cloud computing environment 40, 42. In yet anotherembodiment, the data stores 56A, 56B 34 are an area of non-volatilememory 30 of the server device 22 that are replicated.

As known in the art, cloud computing environments provide ubiquitous,convenient, on-demand network access to a shared pool of configurablecomputing resources (e.g., networks, servers, storage, applications, andservices) that can be rapidly provisioned and released with minimalmanagement effort or service provider interaction. Accordingly, each ofthe cloud computing environments 40, 42 shown in FIG. 2 providecomputational power that can be supplied, utilized, and paid forentirely on demand. Although there are only two cloud computingenvironments 40, 42 shown in the FIG. 2 schematic, it will beappreciated by one skilled in the art that the present invention is notlimited to using two cloud computing environments 40, 42 and may utilizeone or more cloud computing environments. Advantageously, by utilizing aplurality of cloud computing environments, as shown in the FIG. 2example, additional system redundancy and resiliency may be achieved. Byway of example and in no manner limiting, example cloud computingenvironments used with the present invention may include Amazon EC2®,Rackspace® and GoGrid®.

Each of the NOC user interfaces 46A, 46B includes a web call module 48A,48B, respectively, that provides management and control of their virtualmachines. As shown in the FIG. 2 example, in one embodiment, the NOCuser interfaces 46A, 46B are distributed across a number of virtualmachines and clouds sharing a same Domain Name system (DNS).Advantageously, by distributing the NOC user interfaces 46A, 46B acrossvirtual machines sharing the same DNS, a round-robin technique of datastore updates may be achieved. For example, if during an audio webcastone of the virtual machines processing the audio becomes unreachable,connections to a next internet protocol (IP) address in a DNS list ofthe DNS may be made resulting in increased resiliency of the system. Inaddition, in one embodiment, each call module 48A, 48B implementsdatabase replication of the data stores 56A, 56B to ensure thatwhichever virtual machine assumes audio webcast processing, all virtualmachines of the system receive the instructions that have been sent overthe job control signal via link 58 and have information concerning thestate of all virtual machines in the cloud environments 40, 42.Advantageously, by utilizing this technique, any of the web call modules48A, 48B and/or cloud environments 40, 42 may be impaired or disabledwith minimal or no interruption of scheduled or active conferencewebcasts.

For example, referring to the primary cloud environment 40 of FIG. 2,the call module 48A transmits a signal via link 60 to ‘Virtual MachineA’ 44A to dial a Primary Public Switch Telephone Network (PSTN) bridge50A via link 64A (e.g. a PSTN Breakout Service) to connect to the PSTN52. Example PSTN Breakout Services include, but are not limited to,SiPGate®, VoIPTalk™, and VoIPfone™. The Primary PSTN bridge 50A dialsout via a link to the PSTN 50 and returns a SIP audio signal via link64B to the Virtual Machine A 44A. The Virtual Machine A 44A in turntransmits the received audio signal in a streaming encoded format via asignal carried on link 76A to the content delivery network (CDN) 54. Inone embodiment, as discussed in connection with FIG. 3 below, theVirtual Machine A 44A is configured to include a PSTN Bridge/SessionInitiation Protocol (SIP) client that, as shown in FIG. 2 via link 64C,relays the received audio signal to ‘Virtual Machine B’ 44B. VirtualMachine B 44B may then encode and stream the received audio signal in anencoded format via a link 76B to the CDN 54. In the event there is aproblem with Virtual Machine A, the Primary PSTN Bridge 50Aautomatically transmits the SIP audio signal via link 64D to VirtualMachine B 44B, which continues to transmit the encoded audio signal tothe CDN 54 via link 76B.

In one embodiment, if there is a performance issue with the primarycloud environment 40, Virtual Machine B 44B transmits a relay signal vialink 72 of the received audio signal to ‘Virtual Machine C’ 44C in thesecondary cloud computing environment 42. Virtual Machine C 44C thenprocesses the audio signal in a manner similar to Virtual Machine B 44Band transmits/streams the encoded audio signal to the CDN 54 via link76C.

As shown in the FIG. 2 example, in one embodiment, to further ensuresystem availability if there are performance issues with the primarycloud environment 40, the web call module 48B of the secondary cloudcomputing environment 42 transmits a signal via link 68 to ‘VirtualMachine D’ 44D to dial a Secondary Public Switch Telephone Network(PSTN) bridge 50B to connect to the PSTN 52 with the resulting audiobeing made available to the CDN 54 as a completely discrete pathavailable as soon as the NOC User Interface 46B connects the audioconference call to the source conference call.

Advantageously, the system 20 provides failover safety if 1) VirtualMachine A 44A fails to maintain a conference call, 2) the primary cloudenvironment 40 and/or Virtual Machine A 44A fail or have performanceissues, and/or 3) the signal is not transmitted via link 64C fromVirtual Machine A 44A to Virtual Machine B 44B. In particular, if any ofthe before-mentioned situations occur, local failover to Virtual MachineB 44B occurs quickly ensuring there is a continuous audio source. Inaddition, the relay signal transmitted via link 72 from Virtual MachineB 44B to Virtual Machine C 44C ensures that a second virtual machine ina second cloud computing environment is activated, reducing the riskthat faulty clouds may cause an outage.

Moreover, by Virtual Machine D 44D initiating a second call through theSecondary PSTN Bridge 50B, additional redundancy and resiliency may beachieved. While the NOC User Interface 46B may need to clear security tojoin the conference call on the PSTN 52, which may take a few minutes,this action is initiated as soon as Virtual Machine C 44C is activatedas a backup to Virtual Machine B 44B. As such, should Virtual Machine A44A have failed because the primary cloud environment 40 is failing, bythe time Virtual Machine B 44 fails to transmit the signal via link 72to Virtual Machine C 44C, Virtual Machine D 44D is already active andstreaming an encoded audio stream signal via link 76D to the CDN 54,which may be included as an alternative option in a CDN playlist.

Turning now to FIG. 3, a schematic of component modules included in theprimary cloud environment 40 of FIG. 2 is disclosed. As shown in theFIG. 3 example, in one embodiment, a plurality of clone (e.g., replica)virtual machines, indicated by Virtual Machine A 44A and Virtual MachineB 44B, are instantiated by a web call module 48A in the primary cloudenvironment 40. Initially, the plurality of virtual machines 44A, 44Bshares the same initial state.

As shown in FIG. 2, each virtual machine 44A, 44B is configured toinclude a Session Initiation Protocol (SIP) Application 80A, 80B,respectively, having a call handler module 82A, 82B, respectively,configured to dial a PSTN Breakout Service 50. As known in the art, aPSTN Breakout Service provides a platform for transporting Voice overInternet Protocol (VoIP) media between IP networks and a PSTN. In oneembodiment, as shown in FIG. 3, Virtual Machine A 44A establishes acommunication signal via link 92A with the PSTN Breakout Service 50 todial the PSTN to join a conference call. An audio signal of theconference call is then transmitted back to Virtual Machine A 44A overthe communication link 92A. As discussed previously, Virtual machine A44 may also transmit the received audio signal to Virtual Machine B 44Bwhich in turn, transmits an encoded audio stream to the CDN 54. Examplecommunication protocols used for receiving and transmitting audiosignals between each call handler 82A, 82B and the PSTN Breakout Service50 may include, but are not limited to, G.722, G.719, G.711, SPEEX andGSM.

In one embodiment, each of the call handler modules 82A, 82B, uponreceiving digital audio signals, compresses the sound sequence includedtherein into a digital audio encoding format. In one embodiment, thedigital audio encoding format uses a form of lossy data compression,such as an MP3 encoding format. Each call handler module 82A, 82B thentransmits each respective MP3 encoded format file to a data store 100A,100B, respectively, in the primary cloud 40 using file transfer protocol(FTP). In one embodiment, an audio scrub module (not shown) is providedthat may be applied to the MP3 file to improve the quality of the audiofile. Upon completion of the conference call and/or audio filescrubbing, as shown in FIG. 3, each MP3 encoded format file 102A, 102Bmay be archived in archives 104A, 104B, respectively, and thentransmitted via links 114A, 114B, respectively, to the CDN 54 using FTP.

In one embodiment, as shown in the FIG. 3 example, each Virtual Machine44A, 44B may be configured to include a Windows Driver Model (WDM) audiodevice driver 84A, 84B, respectively. Each WDM audio device driver 84A,84B is configured to operate as an Operating System (OS) audio bus thatprovides audio converter and splitter functionality resulting in anaudio card simulation in each Virtual Machine 44A, 44B. As shown in theFIG. 3 example, the received digital audio signal may be encoded into anuncompressed audio format via links 98A, 98B, such as PCM.

Each of uncompressed audio formats 98A, 98B may be then provided torespective Windows Media Format Software Development Kit (WMFSDK)encoders 86A, 86B included in each Virtual Machine 44A, 44B. The WMFSDKencoders 86A, 86B encode the uncompressed audio into an Advanced SystemsFormat (ASF) and transmit/stream the encoded files via links 106A, 106B,respectively, to the CDN 54 using Hypertext Transfer Protocol (HTTP). Asshown in the FIG. 3 example, each Virtual Machine 44A, 44B may alsoinclude a Flash Media Encoder Software Development Kit (FMESDK) 88A, 88Bfor encoding the uncompressed audio format into an encoded flash audioformat. In one embodiment, output audio streams from each FMESDK 88A,88B are transmitted to a Flash Media Development Server (FMDS) 90A, 90Bvia a Real Time Messaging Protocol (RTMP) (e.g., a protocol developed byAdobe Systems, Incorporated, for streaming audio and other types ofmedia). Each FMDS 90A, 90B then streams the flash encoded audio formatsto the CDN 54 using RTMP and RTMP digital video record (DVR) techniques.

Turning now to FIG. 4, an example method of capturing conference callaudio and delivering the conference call audio in streaming mediaformats is disclosed. First, at step 210, a first and second virtualmachine is instantiated in a first cloud computing environment by thewebcall module of the NOC interface. Next, at step 212, a call handlermodule of a SIP application included in a first virtual machineestablishes a connection to a first PSTN Breakout Service. Next, at step214, the call handler module issues a command to the PSTN BreakoutService to join a conference call. Once the first PSTN Breakout Servicejoins the conference call, at step 216, the call handler module receivesa digital audio stream of the conference call. At step 218, the firstvirtual machine then relays the received digital audio stream to thesecond virtual machine. Then, at step 220, the first and the secondvirtual machines encode the received digital audio. Next, at step 222,the first and second virtual machines stream the encoded digital audioin various media formats to a CDN for playing in a browser by an enduser. In one embodiment, the encoded media formats include, but are notlimited to, a Windows Media Audio format, a Flash Audio format, and aMP3 audio format.

As described previously, in one embodiment, each call handler module inresponse to receiving the digital audio from the PSTN Breakout Servicemay generate and transmit a MP3 encoded format file to a data storeusing file transfer protocol (FTP). The MP3 file may then be audioscrubbed by an audio scrub module to improve the quality of the audiofile. Upon completion of the conference call and/or audio filescrubbing, the MP3 encoded format file may be archived and thentransmitted via a link to the CDN using FTP.

In one embodiment, at step 224, the method includes monitoring stateinformation of the first and second virtual machine, as well as thefirst cloud computing environment. The state information may relate toprocessing throughput. In the event of a delay or failure associatedwith the first virtual machine or first PSTN Breakout Service, at step226, the second virtual machine may establish a connection with a secondPSTN Breakout Service to join the conference call, receive and encodethe digital audio, and continue streaming the encoded digital stream tothe CDN.

At step 228, the method may include the webcall module instantiating athird virtual machine in a second cloud computing environment inresponse to state information associated with either the first virtualmachine, second virtual machine, or first cloud computing environment.As discussed previously, additional virtual machines may be establishedin either the first or second cloud computing environments and thepresent invention is not limited to the number of virtual machinesdepicted or described in FIGS. 2-4. In the event a third virtual machineis instantiated in the second cloud computing environment, at step 230,either the first or second virtual machine may transmit their receiveddigital audio stream to the third virtual machine. At step 232, thethird virtual machine encodes the received digital audio and at step234, streams the encoded digital audio from the third machine in thesecond cloud computing environment to the CDN. In one embodiment, asshown at step 236, the third virtual machine may also establish aconnection with a second PSTN Breakout Service to receive the digitalaudio and transmit the received digital audio to additional virtualmachines for added system redundancy and resiliency.

Various features of the system may be implemented in hardware, software,or a combination of hardware and software. For example, some features ofthe system may be implemented in one or more computer programs executingon programmable computers. Each program may be implemented in a highlevel procedural or object-oriented programming language to communicatewith a computer system or other machine. Furthermore, each such computerprogram may be stored on a storage medium such as read-only-memory (ROM)readable by a general or special purpose programmable computer orprocessor, for configuring and operating the computer to perform thefunctions described above.

What is claimed is:
 1. A method of streaming information comprising:establishing a set of connections between a first public switchedtelephone network (PSTN) breakout service and a set of machines, a firstmachine in the set of machines being a first virtual machineinstantiated in a first cloud computing environment, the first PSTNbreakout service being connected to a conference call; receiving adigital data stream in the first virtual machines, the digital datastream having been encoded according to a first encoding protocol,wherein the digital data stream is received over one of a set ofconnections between the first PSTN breakout service and the firstvirtual machine; audio encoding the digital data stream on the firstvirtual machine to generate an encoded digital data stream, the encodingbeing done according to a second encoding protocol; transmitting theencoded digital data stream from the first virtual machine to a contentdelivery network (CDN); receiving the digital data stream encodedaccording to the first encoding protocol at a second virtual machine,and when an event affects a performance of the first virtual machine,audio encoding the digital data stream on the second virtual machine togenerate an encoded digital data stream, the encoding being doneaccording to the second encoding protocol, and transmitting the encodeddata stream from the second virtual machine to the CDN; and providing asession initiation protocol (SIP) application in each of the first andsecond virtual machines for receiving and managing the digital datastream in the first encoding protocol.
 2. The method of claim 1, whereinthe SIP application in the first virtual machine relays the receivedaudio signal to the second virtual machine.
 3. The method of claim 1,wherein the second virtual machine is instantiated in the first cloudcomputing environment or a second cloud computing environment differentfrom the first cloud computing environment.
 4. The method of claim 1,further comprising instantiating the first virtual machine in responseto a request from a call control interface module.
 5. The method ofclaim 4, comprising instantiating the second virtual machine from atleast one of the call control interface module and the first virtualmachine.
 6. The method of claim 1, comprising receiving the digital datastream at the second virtual machine from the first virtual machine. 7.The method of claim 1, comprising receiving the digital data stream atthe second virtual machine from the first PSTN breakout service.
 8. Themethod of claim 1, comprising receiving the digital data stream from asecond PSTN breakout service, the second PSTN breakout service beingconnected to the conference call.
 9. The method of claim 1, comprisingtransmitting the digital data stream to the second virtual machine fromthe first PSTN breakout service upon at least one of a failure and delayin one of the data connections between the first data connection and thefirst virtual machine.
 10. The method of claim 1, wherein the digitaldata stream is received using a speech codec.
 11. The method of claim10, wherein the codec is selected from the group consisting of G.722,G.719, G.711, SPEEX and GSM.
 12. The method of claim 1, furthercomprising providing an operating system (OS) audio bus for transmittingthe digital data stream in each of the first and second virtualmachines, the OS audio bus being a software audio driver.
 13. The methodof claim 12, wherein the software audio driver is a Windows Driver Model(WDM) audio driver.
 14. The method of claim 1, comprising encoding thedigital data stream into a Windows Media Audio (WMA) file.
 15. Themethod of claim 1, further comprising: providing a plurality of audioencoders for encoding the digital data stream into at least one thirdencoding protocol different from the first encoding protocol and thesecond encoding protocol; encoding the digital data stream using thethird encoding protocol; and transmitting the encoded digital datastream according to the third protocol to the CDN.
 16. The method ofclaim 15, wherein a first encoder of the plurality of audio encoders isa Windows Media encoder and a second encoder of the plurality of audioencoders is a Flash Media encoder.
 17. The method of claim 16,comprising encoding the digital data stream into a Flash Media audiofile.
 18. The method of claim 1, further comprising providing an audioserver in each of the first and second virtual machines, the audioserver transmitting the encoded digital data stream to the CDN.
 19. Themethod of claim 18, wherein the audio server is a flash media server andthe encoded digital data stream is a flash media file.
 20. The method ofclaim 1, further comprising: generating, from the SIP Application, anMP3 audio file from the received digital data stream in the firstencoding format; and storing the MP3 audio file in a data storeassociated with each of the first and second virtual machines.
 21. Themethod of claim 20, further comprising audio scrubbing the MP3 audiofile.
 22. A system for streaming information comprising: a callprocessor configured to initiate a set of connections between a firstpublic switched telephone network (PSTN) breakout service and a set ofmachines, a first machine in the set of machines being a first virtualmachine instantiated in a first cloud computing environment, the firstPSTN breakout service being connected to a conference call; the firstvirtual machine including: a receiver configured to receive a digitaldata stream having been encoded according to a first encoding protocol;an audio encoder configured to encode, according to a second encodingprotocol, the digital data stream to generate an encoded digital datastream; a transmitter configured to transmit the encoded digital datastream to a content delivery network (CDN), wherein the first virtualmachine is configured to encode the digital data stream on the firstvirtual machine in response to receiving the digital data stream overone of the set of connections between the PSTN breakout service and thefirst virtual machine, and a transmitter configured to transmit theencoded digital data stream from the first virtual machine to the CDN,and a second virtual machine including: a receiver configured to receivethe digital data stream having been encoded according to the firstencoding protocol; an audio encoder configured to encode, according to asecond encoding protocol, the digital data stream to generate a secondencoded digital data stream, when an event affects a performance of thefirst virtual machine; and a transmitter configured to transmit theencoded data stream from the second virtual machine to the CDN, asession initiation protocol (SIP) application included in each of arespective one of the first and second virtual machines, which whenimplemented by the respective virtual machine, causes the respectivevirtual machine to receive and manage the digital data stream in thefirst encoding protocol.
 23. The system of claim 22, wherein the SIPapplication is configured in the first virtual machine to relay thereceived audio signal to the second virtual machine.
 24. The system ofclaim 22, wherein the second virtual machine is instantiated in thefirst cloud computing environment or a second cloud computingenvironment different from the first cloud computing environment. 25.The system of claim 22, comprising means for instantiating the firstvirtual machine in response to a request from a call control interfacemodule.
 26. The system of claim 25, comprising means for instantiatingthe second virtual machine from at least one of the call controlinterface module and the first virtual machine.
 27. The system of claim22, comprising means for receiving the digital data stream at the secondvirtual machine from the first virtual machine.
 28. The system of claim22, comprising means for receiving the digital data stream at the secondvirtual machine from the first PSTN breakout service.
 29. The system ofclaim 22, comprising means for receiving the digital data stream from asecond PSTN breakout service, the second PSTN breakout service beingconnected to the conference call.
 30. The system of claim 22, whereinthe digital data stream is transmitted to the second virtual machinefrom the first PSTN breakout service upon at least one of a failure anddelay in one of the data connections between the first data connectionand the first virtual machine.
 31. The system of claim 22, wherein thedigital data stream is received using a speech codec.
 32. The system ofclaim 31, wherein the codec is selected from the group consisting ofG.722, G.719, G.711, SPEEX and GSM.
 33. The system of claim 22, furthercomprising an operating system (OS) audio bus for transmitting thedigital data stream in each of the first and second virtual machines,the OS audio bus being a software audio driver.
 34. The system of claim33, wherein the software audio driver is a Windows Driver Model (WDM)audio driver.
 35. The system of claim 22, comprising means for encodingthe digital data stream into a Windows Media Audio (WMA) file.
 36. Thesystem of claim 22, further comprising a plurality of audio encoders inwhich a first encoder of the plurality of encoders encodes the digitaldata stream into at least one third encoding protocol different from thefirst encoding protocol and the second encoding protocol, and transmitsthe encoded digital data stream according to the third protocol to theCDN.
 37. The system of claim 36, wherein the first encoder of theplurality of audio encoders is a Windows Media encoder and a secondencoder of the plurality of audio encoders is a Flash Media encoder. 38.The system of claim 36, wherein the digital data stream is encoded intoa Flash Media audio file.
 39. The system of claim 22, further comprisingan audio server in each of the first and second virtual machines, theaudio server configured to transmit the encoded digital data stream tothe CDN.
 40. The system of claim 39, wherein the audio server is a flashmedia server and the encoded digital data stream is a flash media file.41. The system of claim 22, wherein the SIP Application generates an MP3audio file from the received digital data stream in the first encodingformat, and stores the MP3 audio file in a data store associated witheach of the first and second virtual machines.
 42. The system of claim41, further comprising means for audio scrubbing the MP3 audio file.